Neural noise reduction with linear and nonlinear filtering for single-channel audio signals
Abstract
This disclosure provides methods, devices, and systems for audio signal processing. The present implementations more specifically relate to speech enhancement techniques that combine statistical signal processing with neural network inferencing. In some aspects, a speech enhancement system may include a linear filter, a deep neural network (DNN), and a nonlinear post-filter. The linear filter and the nonlinear post-filter are configured to suppress noise in audio signals using statistical signal processing techniques. More specifically, the linear filter denoises an input audio signal based on a temporal correlation between successive frames of the audio signal. The DNN infers a speech signal and a noise signal (representing a speech component and a noise component, respectively, of the audio signal) based on the denoised audio signal. The nonlinear post-filter suppresses residual noise in the speech signal based on one or more Gaussian mixture models (GMM) associated with the speech signal and the noise signal.
Claims
exact text as granted — not AI-modifiedWhat is claimed is:
1 . A method of speech enhancement, comprising:
receiving a series of frames of an audio signal; denoising a first frame in the series of frames based at least in part on a temporal correlation between the series of frames; inferring a probability of speech associated with the denoised first frame based on a neural network model; generating a first speech signal and a first noise signal based on the probability of speech associated with the denoised first frame, the first speech signal and the first noise signal representing a speech component and a noise component, respectively, of the audio signal in the first frame; determining a first spectral suppression gain based on the first speech signal and the first noise signal; and suppressing residual noise in the first speech signal based on the first spectral suppression gain.
2 . The method of claim 1 , wherein the audio signal comprises a single channel of audio data.
3 . The method of claim 1 , wherein the first frame is denoised based on a multi-frame minimum variance distortionless response (MF-MVDR) beamformer that reduces a power of the noise component of the audio signal without distorting the speech component.
4 . The method of claim 1 , further comprising:
determining an interframe correlation (IFC) vector associated with a speech component of the audio signal based at least in part on the probability of speech associated with the denoised first frame; denoising a second frame in the series of frames based at least in part on the IFC vector; inferring a probability of speech associated with the denoised second frame based on the neural network model; generating a second speech signal and a second noise signal based on the probability of speech associated with the denoised second frame, the second speech signal and the second noise signal representing a speech component and a noise component, respectively, of the audio signal in the second frame; determining a second spectral suppression gain based on the second speech signal and the second noise signal; and suppressing residual noise in the second speech signal based on the second spectral suppression gain.
5 . The method of claim 1 , wherein the speech component of the audio signal in the denoised first frame is equal to the speech component of the audio signal in the first speech signal.
6 . The method of claim 1 , wherein the determining of the first spectral suppression gain comprises:
determining a number (M) of voice activity detection (VAD) features that are indicative of whether speech is present in the first frame based at least in part on the first speech signal and the first noise signal; determining M probabilities of speech associated with the first speech signal based on the M VAD features, respectively; and determining a magnitude or power of the residual noise in the first speech signal based on the M probabilities of speech associated with the first speech signal.
7 . The method of claim 6 , wherein the magnitude or power of the residual noise in the first speech signal is determined based only on the lowest probability of speech among the M probabilities of speech associated with the first speech signal.
8 . The method of claim 6 , wherein each of the M probabilities of speech associated with the first speech signal is determined based on a respective Gaussian mixture model (GMM).
9 . The method of claim 6 , wherein M>1.
10 . The method of claim 6 , wherein the M VAD features include a normalized difference between the first speech signal and the first noise signal.
11 . The method of claim 6 , wherein the M VAD features include at least one of a cepstral peak, a spectral entropy, or a harmonic product spectrum (HPS) associated with the first speech signal.
12 . A speech enhancement system comprising:
a processing system; and a memory storing instructions that, when executed by the processing system, causes the speech enhancement system to:
receive a series of frames of an audio signal;
denoise a first frame in the series of frames based at least in part on a temporal correlation between the series of frames;
infer a probability of speech associated with the denoised first frame based on a neural network model;
generate a first speech signal and a first noise signal based on the probability of speech associated with the denoised first frame, the first speech signal and the first noise signal representing a speech component and a noise component, respectively, of the audio signal in the first frame;
determine a first spectral suppression gain based on the first speech signal and the first noise signal; and
suppress residual noise in the first speech signal based on the first spectral suppression gain.
13 . The speech enhancement system of claim 12 , wherein the audio signal comprises a single channel of audio data.
14 . The speech enhancement system of claim 12 , wherein the first frame is denoised based on a multi-frame minimum variance distortionless response (MF-MVDR) beamformer that reduces a power of the noise component of the audio signal without distorting the speech component.
15 . The speech enhancement system of claim 12 , wherein execution of the instructions further causes the speech enhancement system to:
determine an interframe correlation (IFC) vector associated with a speech component of the audio signal based at least in part on the probability of speech associated with the denoised first frame; denoise a second frame in the series of frames based at least in part on the IFC vector; infer a probability of speech associated with the denoised second frame based on the neural network model; generate a second speech signal and a second noise signal based on the probability of speech associated with the denoised second frame, the second speech signal and the second noise signal representing a speech component and a noise component, respectively, of the audio signal in the second frame; determine a second spectral suppression gain based on the second speech signal and the second noise signal; and suppress residual noise in the second speech signal based on the second spectral suppression gain.
16 . The speech enhancement system of claim 12 , wherein the speech component of the audio signal in denoised first frame is equal to the speech component of the audio signal in the first speech signal.
17 . The speech enhancement system of claim 12 , wherein the determining of the first spectral suppression gain comprises:
determining a number (M) of voice activity detection (VAD) features that are indicative of whether speech is present in the first frame based at least in part on the first speech signal and the first noise signal; determining M probabilities of speech associated with the first speech signal based on the M VAD features, respectively; and determining a magnitude or power of the residual noise in the first speech signal based on the M probabilities of speech associated with the first speech signal.
18 . The speech enhancement system of claim 17 , wherein each of the M probabilities of speech associated with the first speech signal is determined based on a respective Gaussian mixture model (GMM).
19 . The speech enhancement system of claim 17 , wherein M>1.
20 . The speech enhancement system of claim 17 , wherein the M VAD features include at least one of a normalized difference between the first speech signal and the first noise signal, a cepstral peak of the first speech signal, a spectral entropy of the first speech signal, or a harmonic product spectrum (HPS) associated with the first speech signal.Join the waitlist — get patent alerts
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